openKylin opus package

opus 1.3.1-ok1 source package in openKylin

Changelog

opus (1.3.1-ok1) yangtze; urgency=medium

  * Build for openKylin.

 -- openKylinBot <email address hidden>  Mon, 25 Apr 2022 22:03:04 +0800

Upload details

Uploaded by:
openKylinBot
Sponsored by:
Cibot
Uploaded to:
Yangtze V1.0
Original maintainer:
Ubuntu Developers
Architectures:
any all
Section:
sound
Urgency:
Medium Urgency

Publishing See full publishing history

Series Pocket Published Component Section
Yangtze V1.0 release main sound

Downloads

File Size SHA-256 Checksum
opus_1.3.1.orig.tar.gz 1023.7 KiB e7c49abfa28c3a8670f2f1519087ac9c8b0333df82c3fe0fd5be37df48ec6a5d
opus_1.3.1-ok1.debian.tar.xz 4.4 KiB 921cab5b832cdabbb2d9b4eb30e0f78ccce366d89dcc6ef3e0d3aa67267614a7
opus_1.3.1-ok1.dsc 1.8 KiB 3e725ca74196b0783337a90be9d7c4fa4790a2cb1e6a2dbb79be6d0f88ca9e83

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Binary packages built by this source

libopus-dbg: debugging symbols for libopus

 This package provides the detached debug symbols for libopus.

libopus-dev: Opus codec library development files

 The Opus codec is designed for interactive speech and audio transmission over
 the Internet. It is designed by the IETF Codec Working Group and incorporates
 technology from Skype's SILK codec and Xiph.Org's CELT codec.
 .
 It is intended to suit a wide range of interactive audio applications,
 including Voice over IP, videoconferencing, in-game chat, and even remote live
 music performances. It can scale from low bit-rate narrowband speech to very
 high quality stereo music. The current features are:
 .
  Bit-rates from 6 kb/s 510 kb/s
  Sampling rates from 8 to 48 kHz
  Frame sizes from 2.5 ms to 60 ms
  Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
  Audio bandwidth from narrowband to full-band
  Support for speech and music
  Support for mono and stereo
  Support for up to 255 channels (multistream frames)
  Dynamically adjustable bitrate, audio bandwidth, and frame size
  Good loss robustness and packet loss concealment (PLC)
  Floating point and fixed-point implementation
 .
 This package provides the Opus library headers and development files.

libopus-doc: libopus API documentation

 The Opus codec is designed for interactive speech and audio transmission over
 the Internet. It is designed by the IETF Codec Working Group and incorporates
 technology from Skype's SILK codec and Xiph.Org's CELT codec.
 .
 This package contains the developer documentation for libopus.

libopus0: Opus codec runtime library

 The Opus codec is designed for interactive speech and audio transmission over
 the Internet. It is designed by the IETF Codec Working Group and incorporates
 technology from Skype's SILK codec and Xiph.Org's CELT codec.
 .
 It is intended to suit a wide range of interactive audio applications,
 including Voice over IP, videoconferencing, in-game chat, and even remote live
 music performances. It can scale from low bit-rate narrowband speech to very
 high quality stereo music. The current features are:
 .
  Bit-rates from 6 kb/s 510 kb/s
  Sampling rates from 8 to 48 kHz
  Frame sizes from 2.5 ms to 60 ms
  Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
  Audio bandwidth from narrowband to full-band
  Support for speech and music
  Support for mono and stereo
  Support for up to 255 channels (multistream frames)
  Dynamically adjustable bitrate, audio bandwidth, and frame size
  Good loss robustness and packet loss concealment (PLC)
  Floating point and fixed-point implementation
 .
 This package provides the Opus runtime library.